Issues 18000 - 18999

[..]
ASTERISK-18000: If asterisk starts before my Internet connection is up, asterisk cannot register to any of my registrations and reports DNS errors
ASTERISK-18002: Voicemail MWI no longer is sent
ASTERISK-18019: MWI Subscriptions have no effect and are not handled correctly
ASTERISK-18023: Typo in Goto/GotoIf documentation
ASTERISK-18024: Incorrect data in CDRs (billsec >> durration)
ASTERISK-18025: CPU spikes when using timerfd timing
ASTERISK-18026: deadlock every 2 hours
ASTERISK-18027: SMS queued with smsq not sent
ASTERISK-18028: Asterisk 1.8 Realtime provider
ASTERISK-18029: Aastra 480i does not work with Asterisk 1.8
ASTERISK-18031: Deadlock in ast_async_goto() because of wrong locking order
ASTERISK-18032: [patch] - IPv6 and IPv4 NAT not working
ASTERISK-18036: Enhance Automated Dialplan Tests
ASTERISK-18037: [patch] Add hold status with CEL and setting a channel var HOLDING via ast_moh_stop/start and potential res_musiconhold fixup.
ASTERISK-18038: Only one digit on call transfer
ASTERISK-18039: Realtime music restarts from beginning each time
ASTERISK-18040: Asterisk segfaults on shutdown when confbridge is in use
ASTERISK-18041: distribute voicemail to other boxes using database lookup
ASTERISK-18042: [patch] information from unsolicited MWI is lost during asterisk restart; cache to ast_db to avoid this
ASTERISK-18043: MeetMe Not Respecting SetMusicOnHold()
ASTERISK-18044: extenpatternmatchnew breaks ability for caller to exit queue
ASTERISK-18045: gtalk should subscribe to res_stun_monitor
ASTERISK-18046: commit code for 'stun show status'
ASTERISK-18054: Huge list of frozen SIP channels
ASTERISK-18055: Attempting to register to remote server that returns 404 causes no further registration attempts
ASTERISK-18056: Test Issue - Just Ignore
ASTERISK-18058: OOH323 Fails to Identify Dead Peer and Rings Indeffinitely
ASTERISK-18060: Playback in h exten triggers a null pointer in sip_setoptions resulting in a crash
ASTERISK-18061: Asterisk periodically locksup, using CPU time.
ASTERISK-18062: menuselect dependency resolution is broken with gcc 4.6
ASTERISK-18063: Flooding with [Jun 24 19:33:17] WARNING[6995]: chan_sip.c:6213 sip_write: Asked to transmit frame type g726, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw)
ASTERISK-18064: Asterisk's AMI over HTTP adds an extra \r\n in the headers causing strict parsers to fail
ASTERISK-18065: 1.4 fails to build with gcc 4.6 in dev-mode: chan_phone.c:307:9: error: the comparison will always evaluate as 'true'
ASTERISK-18066: Attended transfert with sendrpid=yes and directedmedia=yes with aastra phone, return 500 error and not works
ASTERISK-18067: REGRESSION: After upgrading from 1.4.41 to *1.8.4.2 a SIP extension with a voicemail box can no longer monitor mwi of another extension
ASTERISK-18068: Request to use ConfBridge() instead of MeetMe() in Page()
ASTERISK-18069: [patch] app_queue Add Login Time and Last Paused Times to Queue Members
ASTERISK-18070: Asterisk 1.4.22 - 1.4.35 crashes once or twice a week
ASTERISK-18071: app_queue Add Member StateInterface to Output of "queue show" (CLI) and "QueueStatus" (AMI)
ASTERISK-18072: Segfault when using Directory()
ASTERISK-18073: If INVITE transaction runs in parallel with INFO transaction, 200 OK for INVITE does not contain contact headers
ASTERISK-18074: SIP messages stop being processed
ASTERISK-18076: Asterisk 1.6.1 won't "answer" the phone when using a callcentric sip trunk
ASTERISK-18077: When in queue on g722 with interruptions, music on hold can get stuck and no longer play
ASTERISK-18078: [patch] Segfault when publishing device states via XMPP and not connected
ASTERISK-18079: Flash doesn't break dialtone on DAHDI FXS channels
ASTERISK-18080: DUNDi lookups cause deadlocks
ASTERISK-18081: Queues - Missed calls - SIP cancel reason
ASTERISK-18082: Deadlock of SIP or segfault when doing REFERs
ASTERISK-18083: "r" dial params stop give ringback if M macro used
ASTERISK-18084: chan_gtalk only receives calls from Gtalk Android but not viceversa
ASTERISK-18085: Asterisk crashes when using incorrect Dial() syntax
ASTERISK-18086: Can't place calls on hold with certain IP phones (aastra 9133i and sipdroid soft phone)
ASTERISK-18087: asterisk will crash on "module reload"
ASTERISK-18089: Using comment ;--- causes dialplan corruption
ASTERISK-18090: ERROR[15785]: chan_iax2.c:1742 iax2_getpeername: Bad address cast to IPv4
ASTERISK-18091: Every "sip notify" cmd open a udptl port, and does not free it
ASTERISK-18092: asterisk segfault libpthread-2.9.so
ASTERISK-18093: Add an extra argument to app_originate (dialplan originate()) to state that it should run async
ASTERISK-18094: iLBC (30ms packet) to G711 (20ms) ULAW transcoding sounds "robotic"
ASTERISK-18095: Misleading "Call from 'xxx' to extension 'yyy' rejected because extension not found in context" due to missing SIP domain
ASTERISK-18096: While receiving an AMI event:MeetmeLeave, the Uservalue: has non-ascii values
ASTERISK-18097: svn revision 326678 of the Asterisk Trunk fails to compile on SQLITE3 error
ASTERISK-18100: chan_sip not responding sometimes until asterisk restart
ASTERISK-18101: Asterisk 1.8 Deadlock in app_queue
ASTERISK-18102: Asterisk does not start when compiled and loaded in a Cygwin environment
ASTERISK-18103: asterisk 1.6.2.19 core dump on reload
ASTERISK-18104: asterisk terminates on module load attempt
ASTERISK-18105: most of asterisk modules are unbuildable in cygwin environment
ASTERISK-18106: specifying features.conf to be loaded by static realtime is reported as bound when extconfig.conf is parsed but is ignored
ASTERISK-18107: Asterisk is not answering correctly the OPTIONS messages sent by SIP provider.
ASTERISK-18108: building asterisk under cygwin: errors in main/editline/np/unvis.c
ASTERISK-18109: Segfault in shell_helper in func_shell.c
ASTERISK-18111: chan_iax2 module unbuildable in cygwin environment
ASTERISK-18127: Queues - Missed calls - SIP cancel reason
ASTERISK-18128: It started to crash several times x day 3 to 4 days ago
ASTERISK-18129: Urgent need for IP permit-deny at the GLOBAL level, not at the peer level
ASTERISK-18130: core show application sendfax/receivefax shows XML docs for app_fax version even when res_fax is loaded
ASTERISK-18131: IAX_COMMAND_HANGUP-Frame not sent after Hangup(${HANGUPCAUSE})
ASTERISK-18132: Asterisk segmentation fault (core dump)
ASTERISK-18133: AsteriskGUI consistently crashes Asterisk
ASTERISK-18134: Reload hangs when using func_odbc for hints.
ASTERISK-18135: removal of a specific extension that happens to be a prefix of another extension causes memory corruption
ASTERISK-18136: r319652 causes deadlock with REFER
ASTERISK-18137: Configurations saved in ~/.asterisk.makeopts ignored
ASTERISK-18138: lock.c:280 __ast_pthread_mutex_lock: ccss.c line 3478 (ast_cc_offer): Error obtaining mutex: Invalid argument
ASTERISK-18139: Change libsrtp "not found" message to not suggest --prefix be used
ASTERISK-18140: Expires handling in SUBSCRIBE confuses the absence of the Expires header field with an unsubscribe action.
ASTERISK-18141: "make menuselect" in 1.8.5.0 does not alphabetically sort options
ASTERISK-18142: unresponsive to sip requests
ASTERISK-18143: [patch] Add diversion header to 302 redirect SIP message if we have redirection data
ASTERISK-18144: PickupChan not working correctly
ASTERISK-18145: [regression] With IMAP Voicemail MWI not cleared or set for SIP device
ASTERISK-18146: Unable to register an endpoint after upgrading from 1.4.2.20 to 1.8.3.1
ASTERISK-18147: SIP Phone (Cisco)-> Asterisk 1.4.31 -> IAX-> Asterisk 1.6.2.19 > SIP provider -> PSTN IVR DTMF - DTMF broken after upgrade 1.4 to 1.6 or 1.8
ASTERISK-18149: Dead lock in parking chan_sip handle_request_refer
ASTERISK-18151: Crash in Asterisk under 64 bit arch after authenticating correctly via Asterisk GUI
ASTERISK-18152: [patch] AMI AgentCalled CallerID is Agent extension/name
ASTERISK-18153: Wrong Caller ID Sent in Invite From Header From Realtime SIP User
ASTERISK-18154: Regression: Channel Restarts, Congestion, Limited Call Capacity on PRI
ASTERISK-18155: bridge_softmix.c line 149 (softmix_bridge_leave): Error: attempt to destroy invalid mutex '&sc->lock'
ASTERISK-18156: SIP messages stop being processed with res_timing_dahdi
ASTERISK-18160: Delay between sip unregistration and peer status change
ASTERISK-18161: res_fax.conf: crash if invalid
ASTERISK-18162: Asterisk manager (AMI) Redirect = No CDR written
ASTERISK-18163: DeadLock in asterisk manager interface
ASTERISK-18164: MixMonitor - Handle is not released before HangUp
ASTERISK-18165: sms sending does not work
ASTERISK-18166: Deadlock: asterisk isn't responding to any sip package anymore
ASTERISK-18167: Functions and modules fail to load
ASTERISK-18168: WIkibot not respecting argsep parameter in parameter tags.
ASTERISK-18169: Asterisk crashes on incoming call
ASTERISK-18170: Asterisk 1.8 bri cards (b410p or openvox ) libpri
ASTERISK-18171: Ajax Bug Asterisk 1.8 Manager using mxml
ASTERISK-18172: SendDTMF with duration
ASTERISK-18173: Asterisk sip stack crashing
ASTERISK-18174: Couldn't execute statment: SQL logic error or missing database
ASTERISK-18193: Asterisk doesnt work
ASTERISK-18194: Asterisk 1.6. No audio when using MixMonitor on sip channels
ASTERISK-18195: CLI output is limited to 24000 bytes, running 'dialplan reload' in verbose mode
ASTERISK-18196: Asterisk high cpu usage
ASTERISK-18197: Deadlock - Periodic Deadlock on Transferring Calls
ASTERISK-18199: [regression] LIMIT_CONNECT_FILE does not get played to caller when using dial() app
ASTERISK-18200: CLI output is limited to 24000 bytes, running 'dialplan reload' in verbose mode
ASTERISK-18201: Asterisk should fall back to AVP when SRTP module is not loaded and both SAVP and AVP have been offered
ASTERISK-18202: deadlock in sip
ASTERISK-18203: Problems with NAT on realtime peers (and maybe static ones)
ASTERISK-18204: Mute All Participants
ASTERISK-18205: Deadlock in app_queue when loading real-time queues and handling state change.
ASTERISK-18206: CLONE - chan_gtalk only receives calls from Gtalk Android
ASTERISK-18207: externnotify script called with (null) context parameter during pollmessages run, essentially stopping it from running.
ASTERISK-18208: logger doesn't append data to queue_log file after asterisk starts
ASTERISK-18209: Dialing an incomplete number via dahdi causes a timeout
ASTERISK-18210: chan_iax2 Asterisk crashes after client "crash"
ASTERISK-18211: Asterisk deadlock
ASTERISK-18212: [patch] German saydigits algorithm inserts spurious "and"
ASTERISK-18213: asterisk (1.6.2.19 and 10.0.0-beta1) don't compile on OSX 10.7 (Lion)
ASTERISK-18214: MixMonitor not parsing string variables
ASTERISK-18215: asterisk stop response on SIP messages
ASTERISK-18216: Global variables not set after restart. Must reload everytime.
ASTERISK-18217: [request] Easy generation of AOC (advice of charge) messages
ASTERISK-18218: 10beta1 ooh323 outbound call doesn't work
ASTERISK-18219: t38 gateway doesn't work with ooh323
ASTERISK-18220: MixMonitor stops recording during attended Transfer
ASTERISK-18221: Connecting to my prosody xmpp server crashes res_jabber.so with pkt->from = NULL
ASTERISK-18222: Pickupchan of a local channel segfaults if 2 users pickup at same time
ASTERISK-18223: using http manager commands causes global file descriptor instability, crashing Asterisk
ASTERISK-18224: CDR(accountcode) not accessable to 'Local' channels
ASTERISK-18225: SIP channels are getting stuck after picking up calls
ASTERISK-18226: ConfBridge with two participants passes hold state to the other party. If a 3rd one joins that party stays deaf
ASTERISK-18227: Manual module loading (with global symbols) broken since 1.4.28
ASTERISK-18228: dial() app option r playback early media
ASTERISK-18230: sometimes dialplan switches disappear when merging contexts between pbx_lua and pbx_config
ASTERISK-18231: asterisk crashes when doing multiple ConfBridge
ASTERISK-18232: Broken REGISTER sent to IPv4 server when bindaddr=[::]
ASTERISK-18233: Asterisk does not continue AGI code execution after the channel is hungup by caller.
ASTERISK-18234: IAX2 wont start ring back
ASTERISK-18235: IAX2 wont start ring back early media
ASTERISK-18236: Segmentation fault after second chan_h323 unload
ASTERISK-18237: chan_h323 fails to determine IP/resolve host if host parameter is IP
ASTERISK-18238: local channel doesn't use language of requestor channel
ASTERISK-18239: crash in 1.8
ASTERISK-18240: UniqueID is posted twice to CDR when CDR(userfield) is extended/broken in the process.
ASTERISK-18243: VoiceMail application fails to assign some DTMF codes for application exit when using d() option with context
ASTERISK-18244: When caller hangs up on AGI dial app the code does not complete execution
ASTERISK-18245: Forwarding an Urgent voicemail through VoiceMail to a mailbox that has not been created fails
ASTERISK-18246: When leaving a voicemail marked as Urgent with forwarding recipients, the forward_urgent_auto flag is not respected
ASTERISK-18247: Asterisk taking up 673 mb of ram in idle state
ASTERISK-18248: SIP crash but Asterisk stay UP
ASTERISK-18249: Deadlocks when using a switch statement
ASTERISK-18250: Resource leak in timerfd
ASTERISK-18251: ConfBridge CLI Output Not As Clean As MeetMe
ASTERISK-18252: queue_log mysql time column data format
ASTERISK-18253: Init service isn't compatible with LSB
ASTERISK-18254: say_digits() app sound distortion
ASTERISK-18255: [regression] Non-portable SQL added to app_voicemail
ASTERISK-18257: Develop automated test for SQL portability in Asterisk
ASTERISK-18259: Asterisk 1.8.5.0 not putting p-asserted into ringing or answered SIP messages
ASTERISK-18261: CLONE -asterisk does not starts
ASTERISK-18263: sip directmedia nonat handling / unreachable code
ASTERISK-18264: [patch] Generate security events in chan_sip using new Security Events Framework
ASTERISK-18265: CLONE - [patch] Memory Leak in app_queue
ASTERISK-18266: it should be posible to authenticate sip devices using name different than the section header
ASTERISK-18267: Channel locks after 'core restart now' and 'core reload'
ASTERISK-18268: Improve Menuselect and Asterisk CLI for Module Support Status and CLI
ASTERISK-18269: Calls getting stuck when dialing *8
ASTERISK-18270: Useless message pops every time there is a bridging
ASTERISK-18271: Pattern matching with res_config_mysql extensions does not behave as expected
ASTERISK-18272: Segfault when using HTTP Digest Auth when accessing /amanager
ASTERISK-18273: Orphaned channels after pickup
ASTERISK-18275: DTMF blind transfer continues in dialplan after transfer.
ASTERISK-18277: Memory leak in chan_sip / realtime_peer() / mysql
ASTERISK-18278: cdr_adaptative_odbc does not write CHAR fields
ASTERISK-18279: on-demand recording (*1) filename not being generated correctly due to incorrect uniqueid in filename
ASTERISK-18280: astcanary does not get started when asterisk is startetd with -U <user>
ASTERISK-18281: ERROR[3385]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x0 for 0x11f76860
ASTERISK-18282: Voicemail will not honor forcename / forcegreetings if password is changed and user hangs up
ASTERISK-18283: Call Completion CallCompletionRequest No Core Instance
ASTERISK-18284: Queue remove member command will not accept arguments diplayed by tab complete
ASTERISK-18285: Call Completion CallCompletionRequest No Core Instance WHEN call-limit option set to 1 in sip.conf
ASTERISK-18286: [patch] 'Silence' is truncated in Record()
ASTERISK-18287: Processing in sig_pri.c corrupts memory
ASTERISK-18288: peer->mvipvt needs locking
ASTERISK-18289: FreeBSD: AsteriskUnitTests /main/netsock2/parsing failing
ASTERISK-18290: Mac OSX: Fails to install Asterisk
ASTERISK-18291: Sequence number roll over in res_rtp_multicast.c
ASTERISK-18292: Release Asterisk 1.8.6.0-rc2
ASTERISK-18293: Merge: ASTERISK 18109
ASTERISK-18294: Merge: ASTERISK 18290
ASTERISK-18295: Merge: ASTERISK 18289
ASTERISK-18297: Realtime Asterisk Voice Mail - Sendvoicemail=yes option
ASTERISK-18298: (Call Completion / SIP) SUBSCRIBE fails using TLS
ASTERISK-18299: Merge: ASTERISK 18082
ASTERISK-18300: Merge: ASTERISK 18166
ASTERISK-18301: Outgoing calls fail in chan_gtalk
ASTERISK-18302: System Deadlock, No calls inbound or outbound
ASTERISK-18303: Problem with batch-creation of astdb entries
ASTERISK-18304: Problem with batch-creation of astdb entries
ASTERISK-18305: Problem with batch-creation of astdb entries
ASTERISK-18306: Problem with batch-creation of astdb entries
ASTERISK-18307: Problem with batch-creation of astdb entries
ASTERISK-18308: Problem with batch-creation of astdb entries
ASTERISK-18309: Termination of active call after 20 seconds, because of "Maximum retries exceeded on transmission"
ASTERISK-18310: Termination of active call because of "Maximum retries exceeded on transmission"
ASTERISK-18311: Termination of active call because of Maximum retries exceeded on transmission
ASTERISK-18312: Termination of active call
ASTERISK-18313: termination_of_active_call
ASTERISK-18314: termination_of_active_call
ASTERISK-18315: Problem with batch-creation of astdb entries
ASTERISK-18316: Merge: ASTERISK 18301
ASTERISK-18317: Locking problems with unloading/loading chan_dahdi
ASTERISK-18318: Suspect deadlock in res_ais
ASTERISK-18319: [patch] Optimize chan_sip.c check_rtp_timeout() function
ASTERISK-18320: Outgoing calls fail in chan_gtalk redux
ASTERISK-18321: dynamic_exclude_static option with (temporary) unreachable DNS cause the abend
ASTERISK-18322: ooh323 , alternate gatekeeper
ASTERISK-18323: Memory leak in lock.h
ASTERISK-18324: Kill the last user of a meetme at exit
ASTERISK-18325: "Asked to transmit frame type" slows down all the calls
ASTERISK-18326: Bugs in mISDNuser for NT-PTMP mode causes loss of avaliable procids
ASTERISK-18327: [patch] Monitoring own ip with res_stun_monitor fails when local ip changes
ASTERISK-18328: Asterisk locked
ASTERISK-18330: IAX2 trunk audio problems
ASTERISK-18331: app_sms failure
ASTERISK-18332: chan_dahdi does not seem to report User Rate from Bearer Capabiities
ASTERISK-18333: Chanspy g() option seem to now need precise SPYGROUP name
ASTERISK-18334: chan_dahdi doesn't reset B-channels after calls.
ASTERISK-18335: configure fails if there's a space in the current dir
ASTERISK-18336: chan_vpb: build warnings with gcc 4.6
ASTERISK-18337: chan_h323: warnings when built with gcc 4.6
ASTERISK-18338: DESTDIR does not work when directory has spaces
ASTERISK-18339: Regression: Loss of DTMF signals through asterisk
ASTERISK-18340: CLONE - directmedia or reinvite not working when calling extension that's located an a different asterisk node
ASTERISK-18341: REGEX function documentation
ASTERISK-18342: close() before SSL_Shutdown() in ssl_close()
ASTERISK-18343: extenpatternmatchnew fails to correctly respect dialplan order (regex match before exact in an included context)
ASTERISK-18344: The SIP message "... rejected because extension not found in context ..." lacks vital remote endpoint information
ASTERISK-18345: [patch] sips connection dropped by asterisk with a large INVITE
ASTERISK-18346: MusicOnHold has extra unref which may lead to memory corruption and crash
ASTERISK-18347: Configure --with-imap fails to handle relative paths
ASTERISK-18348: Voicemail with IMAP support cannot be compiled under dev-mode
ASTERISK-18349: Asterisk Crash, with backtrace
ASTERISK-18350: meetme join causes spontaneous Polycom phone reboot
ASTERISK-18351: dnsmgr sets port to 0 after a failed DNS lookup
ASTERISK-18352: add verbose level to logger
ASTERISK-18353: SIP registration doesn't use round-robin DNS.
ASTERISK-18354: sqlite crash for realtime action if config_table is not set
ASTERISK-18355: sqlite realtime_multi_func wrongly assumes commented column exists
ASTERISK-18356: chan_sip realtime_peer has several memory leaks
ASTERISK-18357: chan_dahdi does not compile with --enable-dev-mode and gcc 4.6
ASTERISK-18358: res_jabber does not compile with --enable-dev-mode and gcc 4.6
ASTERISK-18361: http manager getconfig crashes on reading large files/categories (50+ lines)
ASTERISK-18362: AEL: jump doesn't work as 'jump +123456789;'
ASTERISK-18388: no meetme recording / file.c:1222 ast_writefile: No such format '' error
ASTERISK-18389: non-compliant code in chan_sip could be removed for asterisk10 release
ASTERISK-18390: [patch] New DUNDi cli commands to list cache entries
ASTERISK-18391: Asterisk crashing in SQLAllocHandle when using ODBC
ASTERISK-18392: Segmentation fault on Caller ID pattern matching when Caller ID is empty
ASTERISK-18393: Asterisk 1.8.7.0 Blockers
ASTERISK-18394: T.38 FAX passthrough does not work
ASTERISK-18395: Lua applications argument length limitations
ASTERISK-18396: [patch] - Variables Truncated When Using Realtime Dialplan
ASTERISK-18399: When using autoload=yes in modules.conf, res_odbc_conf.so complaints about undefined symbol
ASTERISK-18400: RTCP Receiver Reports are sent for idle RTP sessions
ASTERISK-18401: Debugging messages generated by 'udptl debug' are incomplete
ASTERISK-18402: Asterisk accepts a re-INVITE to switch from T.38 back to voice, but does not switch back
ASTERISK-18403: transfer start ignore digittimout if ! in dialplan
ASTERISK-18404: out-of-order RTP causes DTMF loss
ASTERISK-18405: PRI channel becomes unavailable
ASTERISK-18408: SIP channels stuck
ASTERISK-18409: [patch] /var/lib/asterisk/moh is no longer created by default if no moh files are selected at build time
ASTERISK-18410: Defining same SIP device as user/friend and peer
ASTERISK-18411: Queue members with hints for state_interface get stuck in "In Use" state.
ASTERISK-18412: iLBC issues during install
ASTERISK-18413: chan_misdn has a most broken round robin routiune
ASTERISK-18414: Asterisk DeadLocks After Few Hours of work
ASTERISK-18415: asterisk 1.8 with 99% CPU usage (Meetme with Moh)
ASTERISK-18416: [patch] Realtime queue agents unavailable via AMI before a call event.
ASTERISK-18417: app_alarmreceiver hanging forever in send_tone_burst/ast_waitfor()
ASTERISK-18418: [branch] Set channel variables when manager originates a call
ASTERISK-18419: ERROR[1902] utils.c: write() returned error: Broken pipe
ASTERISK-18420: [regression] Inbound ISDN Overlap dial breaks with Asterisk Version >= 1.8.4.0
ASTERISK-18422: Warning in CLI that seems wrong. (Asked to transmit frame type ulaw, while native formats is 0x100 (g729))
ASTERISK-18423: Crash in AMI initiated sip show peers
ASTERISK-18424: TestSuite: Framework: Allow tests to be dependent on build options
ASTERISK-18425: TestSuite: Framework: Streamline SIPp / pjsua integration with TestClass
ASTERISK-18426: TestSuite: Framework: Add test execution modes
ASTERISK-18427: TestSuite: Framework: Add ability for tests to be executed in subsets
ASTERISK-18428: TestSuite: Framework: Add pre and post test check framework
ASTERISK-18429: TestSuite: Framework: Add pre / post check for memory usage
ASTERISK-18430: TestSuite: Framework: Add pre / post check for threads
ASTERISK-18431: TestSuite: Framework: Add pre / post check for locks
ASTERISK-18432: TestSuite: Framework: Add pre / post check for active channels
ASTERISK-18433: TestSuite: Framework: Add pre / post check for active processes
ASTERISK-18434: TestSuite: Framework: Add pre / post check for SIP dialogs
ASTERISK-18435: TestSuite: Framework: Add pre / post check for file descriptors
ASTERISK-18437: TestSuite: Framework: Add post-test data collection and analysis
ASTERISK-18438: TestSuite: Framework: Add post-test analysis for SIP traffic
ASTERISK-18439: TestSuite: Framework: Add post-test analysis for Asterisk core dump
ASTERISK-18440: TestSuite: Framework: Add generic packet capture option for testsuite
ASTERISK-18441: TestSuite: AstDB: Test Plan Development
ASTERISK-18442: AstDB: Upgrade warning and instructions
ASTERISK-18443: Develop a tool to migrate an SQLite3 AstDB back to Berkley DB
ASTERISK-18444: TestSuite: ConfBridge: Test Plan Development
ASTERISK-18445: TestSuite: Codecs: SIP test
ASTERISK-18446: chan_sip rtcachefriends=no loads fullcontact, but doesn't store it, except in astdb
ASTERISK-18447: Debug manager actions in the CLI
ASTERISK-18450: Should there be transcoding after attended transfer
ASTERISK-18453: manager.c: HTTP Manager, fdopen failed: Bad file descriptor!
ASTERISK-18454: Option for Read to be able to accept #
ASTERISK-18455: RTCP stats works only when transcoding
ASTERISK-18457: Crash in timing.c:169 while sound is being played in ConfBridge
ASTERISK-18479: ast_manager_register_struct attempts to unlock an uninitialized rwlock
ASTERISK-18480: Linear queue orders real time members alphabetically by their interface.
ASTERISK-18487: Daily deadlock issue
ASTERISK-18488: The "pin" parameter of Meetme cmd seems broken.
ASTERISK-18489: Asterisk 10 Beta - T38 NAT not working
ASTERISK-18490: [patch] res_rtp_asterisk.c: ast_rtp_read: Bad address cast to IPv4
ASTERISK-18491: Deadlock on chan_sip / MASTER_CHANNEL
ASTERISK-18492: Deadlock in channel.c / chan_sip.c
ASTERISK-18493: One size sound
ASTERISK-18494: Whisper disconnects call
ASTERISK-18495: module unload chan_iax2.so cause mutex errors
ASTERISK-18496: 1.8.7.0-rc1 breaks dahdi
ASTERISK-18497: Set default tonezone for SIP devices
ASTERISK-18499: Asterisk 1.8.8.0 Release Blockers
ASTERISK-18528: Core Dump on r335064 during system reload.
ASTERISK-18529: Badly needed function strreplace (issue 0018023) needs to be ported to 1.8
ASTERISK-18530: improper use of host LDAP attribute value as ToHost sip client value
ASTERISK-18531: Asterisk 1.8.6 - MySQL Realtime - "language" variable is useless
ASTERISK-18532: Asterisk 10.0.0-beta2 Blockers
ASTERISK-18533: sip channel not closed properly
ASTERISK-18535: [regression] Asterisk 1.8.7.0-rc1: configure error (libpri related)
ASTERISK-18541: Crash under heavy load
ASTERISK-18542: Meetme support have support level 'core'
ASTERISK-18543: Apparent Deadlock in chan_sip continues, even after repeated efforts.
ASTERISK-18544: Pure 1.8 from today. It crashes
ASTERISK-18545: System can crash when using long strings with STRREPLACE()
ASTERISK-18546: Receive WARNING[28319] res_odbc.c: Limit should be a number, not a boolean: '0'. Disabling ODBC class 'db_name'.
ASTERISK-18554: CLI 'manager show command challenge' output missing a required field
ASTERISK-18556: Call parking causes announcment and ringback to caller channel
ASTERISK-18557: Call Parking parkinghints = yes doesn't work as expected
ASTERISK-18558: Option for disabling password less logins to voicemail
ASTERISK-18559: rtptimeout not working per peer
ASTERISK-18560: Crash while executing macro with CALLERID(num) is empty
ASTERISK-18562: Call Parking Ringback Happens to Caller Channel, Rather Than Parking Party
ASTERISK-18565: Voicemail saycid: Play Callers name for external callers if available.
ASTERISK-18566: G.729 RTP Payload Size
ASTERISK-18567: app_queue does not add a cdr it should as it is establishing calls
ASTERISK-18568: Extend the use of Wait to intergrate with res_fax and detect fax/voice
ASTERISK-18569: Extend the use of Wait to intergrate with res_fax and detect fax/voice
ASTERISK-18570: Crashes in RTCP handling
ASTERISK-18571: "core show channel" CLI command blocks the channel during output
ASTERISK-18572: SIP REGISTER fails if :port appears in the To: header
ASTERISK-18573: Lock not released
ASTERISK-18574: SendURL always waits for acknowledgement
ASTERISK-18575: MySQL is detected on RHEL/CentOS when devel libs aren't installed.
ASTERISK-18576: ./configure does not pick up missing mysql dev library files
ASTERISK-18577: National prefix inserted even when caller ID not available
ASTERISK-18578: Asterisk defaults to s@default in pbx_start if extension is not found
ASTERISK-18583: 'r' option to Dial() not working as documented
ASTERISK-18584: SIP Call-ID for B-leg for non-bridged calls
ASTERISK-18585: Dial() Limit reminds callers at the wrong time than that specified in the L option
ASTERISK-18586: When Dial() plays warning messages in the LIMIT options, it put the other party into complete silence
ASTERISK-18587: Enable strictrtp by default
ASTERISK-18588: Static queue agent penalty not respected (rrmemory)
ASTERISK-18593: AEL for loops use Macro app and pipe delimiter
ASTERISK-18602: Voicemail does not read callID from envelope
ASTERISK-18603: SIP Channels not passing DTMF Tones properly
ASTERISK-18604: Constant Lockups throughout the day
ASTERISK-18608: Asterisk 10.0.0-rc1 Blockers
ASTERISK-18609: When sending fax from Cisco 1751V with t.38, after sending re-INVITE asterisk fully ignore SIP messages from Cisco.
ASTERISK-18610: ERRORs since changeset 336294 (Fix bad RTP media bridges)
ASTERISK-18611: RTP packets being repeated and random sequence numbers are being skipped
ASTERISK-18612: Asterisk logs "Audio is at 5060"
ASTERISK-18613: Deadlock (SIP not responding anymore)
ASTERISK-18614: Set Codec for MulticastRTP channel
ASTERISK-18615: Asterisk Randomly Crashes
ASTERISK-18616: Call Forward executes callers channel in wrong context
ASTERISK-18617: ast_srtp_unprotect: SRTP unprotect: authentication failure
ASTERISK-18618: IAX2 needs to be converted to use bitmask lists for codec selections
ASTERISK-18619: Segfault when executing 'core show locks' (debug version)
ASTERISK-18620: Asterisk eating more CPU after upgrade
ASTERISK-18626: The patch found in r333265 on res_jabber.c breaks authentication to a jabber server.
ASTERISK-18627: Music on Hold does not play mp3s until after res_musiconhold.so is reloaded manually.
ASTERISK-18632: Limit PRI channel re-selection to same group in span
ASTERISK-18634: Create VM_INFO() dialplan function to gather information about a mailbox
ASTERISK-18635: CallerID is Reverse name vs cid
ASTERISK-18636: just testing to make a private issue
ASTERISK-18637: 'Maxforwards' appears twice in a 'SipShowPeer' AMI action response
ASTERISK-18638: Crash when using chan_unistim
ASTERISK-18639: Multiple Bridge Events Triggered Upon DTMF Key-Presses
ASTERISK-18640: ${SIP_HEADER(Subject)} does not get anything
ASTERISK-18641: T.38 Passthrough Broken
ASTERISK-18642: asterisk-10.0.0-beta2 , can not run make menuselect
ASTERISK-18643: [patch] Allow sip devices to have externaddr setting
ASTERISK-18644: [branch] Add support for early media in AMI action originate
ASTERISK-18645: There is no difference in queue log between adding member as paused and unpaused.
ASTERISK-18646: App Dial using Option F: Proceed with dialplan execution at the next priority in the current extension if the source channel hangs up.
ASTERISK-18647: Can't open XML documentation
ASTERISK-18648: DAHDI channel causes Asterisk to segfault crash due to unhandled ast_read() NULL return
ASTERISK-18649: SIP-Use-Reason-Header field badly formatted in the SIPShowPeer AMI call response
ASTERISK-18650: Asterisk hangs after failed directed call pickup attempt, logs show "Fixup failed on channel SIP/xxx, strange things may happen."
ASTERISK-18651: Compile failure Debian/sparc64
ASTERISK-18652: Asterisk Doesn't Release RTP ports as it should.
ASTERISK-18653: Parking Slot Number Not Being Cleared
ASTERISK-18659: If connection address in SDP content equals "sent-by" address of the Via header, then send RTP media to same address as SIP responses
ASTERISK-18660: ooh323 does not compile in latest 1.8
ASTERISK-18661: CLONE - chan_gtalk only receives calls from Gtalk Android
ASTERISK-18662: Member penalty ignored because wrong queue membercount
ASTERISK-18663: BLF Subscriptions Causes SIP Deadlock
ASTERISK-18669: SIP Peer Name case not updating on sip reload
ASTERISK-18670: documentation for STAT function is a little sparse
ASTERISK-18671: a new Invite after 5 mins
ASTERISK-18672: Busylevel doesn't appear to limit calls
ASTERISK-18673: Large number of active sip dialogs INVITE in the output "sip show channels".
ASTERISK-18674: CLONE - Flooding with chan_sip.c:6047 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x
ASTERISK-18675: SendFax T.38 don't work
ASTERISK-18676: Update thirdparty mISDN from v1.1.9.1 to v1.1.9.2.
ASTERISK-18677: Update mkrelease and prep_tarball scripts to pull pre-exported documentation
ASTERISK-18678: Option 'g([[context^]exten^]priority)' for Dial Application
ASTERISK-18679: Wrong autopause behavior
ASTERISK-18680: CHANGES and/or UPGRADE.txt files need updating to reflect codec support outside chan_sip
ASTERISK-18682: Voicemail video "crash" when key is pressed
ASTERISK-18683: Update wikibot to export Asterisk 10 command reference
ASTERISK-18684: Export PDF and HTML files for Asterisk 10 from the Asterisk wiki
ASTERISK-18685: Half Attended Transfers ("Blind Transfers") Fail for No Apparent Reason
ASTERISK-18687: CLONE - [regression] Asterisk 1.8.7.0-rc1: configure error (libpri related)
ASTERISK-18688: Reloads with large dialplan cause peers to go lagged
ASTERISK-18689: Error in documentaion of Application_AGI
ASTERISK-18690: Unable to enable a disabled module
ASTERISK-18691: messages WARNING[XXXX] features.c: Failed to play transfer sound! and next the agent goes to log out
ASTERISK-18692: T38 asterisk Answer 488
ASTERISK-18693: "rtp set debug ip" not work
ASTERISK-18694: Chan_Unistim.c error during compilation
ASTERISK-18695: CLONE - Asterisk Crash when Realtime LDAP extensions not found
ASTERISK-18696: peer_iphash_cb empty address on sip reload
ASTERISK-18697: [minivm] Crash in MinivmNotify
ASTERISK-18698: Asterisk crashes withou error
ASTERISK-18699: CDR record not updated when using func_callerid
ASTERISK-18700: chan_sip.c and tcptls.c - possible double close of file descriptor
ASTERISK-18701: Sorting of core show channels verbose
ASTERISK-18702: Improvement of overlap dialling handling in chan_sip
ASTERISK-18703: NO acceptance of SDP packets with set i= field
ASTERISK-18704: Asterisk deadlock
ASTERISK-18705: Asterisk Support of SipConnect 1.1
ASTERISK-18706: UDPTL fail while using directmedia
ASTERISK-18707: QueueSummary event returns incorrect LoggedIn value
ASTERISK-18708: func_curl hangs channel under load
ASTERISK-18709: lua socket.http crashes asterisk
ASTERISK-18710: Alarms not properly set on PRI trunks at startup
ASTERISK-18711: Advance Call Routing Capability.
ASTERISK-18712: [patch] Advance Call Routing Capability
ASTERISK-18713: Autoservice thread is orphaned in a blind transfer during callparking
ASTERISK-18714: Outgoing calls fail again with Google Voice
ASTERISK-18715: Allow dialplan to know in advance about estimated wait time for a caller before sending him in a Queue
ASTERISK-18716: file.c:1352 waitstream_core: Unexpected control subclass '32' after upgrading to asterisk 10.0.0-beta2
ASTERISK-18717: Improve error message for app_confbridge
ASTERISK-18719: res_jabber segfault when using function JABBER_RECEIVE with no message (as when receiving buddy typing notifications)
ASTERISK-18720: chan_sip stops frquently working (deadlock)
ASTERISK-18721: 603 decline is busy not circuit-busy
ASTERISK-18722: ast_expr2 reports "op_times: overflow" on some calculations, though the number is calculated correctly.
ASTERISK-18723: HANGUP agi message does not show up properly in "agi set debug on" output
ASTERISK-18724: crash in __ao2_ref_debug
ASTERISK-18725: Several patches related to the internal editline libraries
ASTERISK-18726: CDR processing appears to hang during channel hangup update
ASTERISK-18727: Invalid extension protection upon adding queue member
ASTERISK-18728: Segfault in app_stack.so on Solaris
ASTERISK-18729: md5secret can not be used with register strings
ASTERISK-18730: Asterisk crashes and there is asterisk Failed to start PBX :( in logfile
ASTERISK-18731: [patch] DUNDi weight parameter not processed correctly
ASTERISK-18732: T38 gateway : CSI / DSR DSI not sent in reply of remote fax CSI / DSR DSI
ASTERISK-18733: No CDR for AMI redirect
ASTERISK-18734: Asterisk crash in click-to-call scenario (SIP only)
ASTERISK-18735: Asterisk spontaneous reboot
ASTERISK-18736: Do not retransmit DTMF signal from INFO oriented mode SIP channel to RFC2833 oriented SIP channel
ASTERISK-18737: AGI Park command attempts to double park the call.
ASTERISK-18738: 1.8.8.0-rc2: chan_h323 no longer built by default
ASTERISK-18739: Endianness Problem with Playback WAV Audio on an Big Endian Proccessor ( format_wav.c)
ASTERISK-18740: Deadlock in queues during dialplan reload
ASTERISK-18742: PRI Span: 1 !! Unknown IE 128 (cs0)
ASTERISK-18743: Asterisk Crash with host unknown
ASTERISK-18744: Asterisk doesn't build on OSX
ASTERISK-18745: http problem when asking for listcommands
ASTERISK-18746: RFC2833 stops responding
ASTERISK-18747: Deadlock in chan_sip / event on send mwi / unsubscribe
ASTERISK-18748: channel ooh323 hangs up calls after about 1 minute
ASTERISK-18749: (Only) First attempt to put a call on hold fails when using SRTP
ASTERISK-18750: crash on parking a call
ASTERISK-18751: parallel build fails when cleantest calls clean
ASTERISK-18752: Problems of Text Messaging in Asterisk 10
ASTERISK-18753: Asterisk crash when using cdr_adaptive_odbc and sql server isn't reacheable
ASTERISK-18754: Queues ringinuse=yes does not ring busy extension
ASTERISK-18755: SDP DTMF negotiation issue: fmtp:101 0-16
ASTERISK-18756: Asterisk crashed randomly - moh
ASTERISK-18757: mohmp3 crashes
ASTERISK-18758: CLONE - Asterisk ignoring sendonly SDP generated from Cisco UCM after generating inactive SDP when a Cisco phone initiates hold
ASTERISK-18759: Asterisk re-uses stale nonce in edge case
ASTERISK-18760: Deadlock in SVN 1.8 version 342359
ASTERISK-18761: Create a new hint type for voicemail boxes
ASTERISK-18762: dialplan remove include without arguments crashes asterisk
ASTERISK-18763: Different behaviour in case of usage of "#include file" instead of direct config pars
ASTERISK-18764: Asterisk stopped accepting sip registrations, and stopped logging to full.
ASTERISK-18765: Memory Leak in lock.h
ASTERISK-18766: Very poor performance when manager.conf enabled=yes
ASTERISK-18800: Sending callerid(name) on PRI may cause call rejection
ASTERISK-18801: Menuselect Easter Egg (Motherships)
ASTERISK-18802: Adds barriers to the menuselect easter egg
ASTERISK-18803: [patch] ast_indicate(chan, -1) don't stop playing tones
ASTERISK-18804: WaitExten(...,m(MOH)) doesn't play the correct audio when Set(CHANNEL(musicclass)=...) is used
ASTERISK-18805: Remote crash vulnerability in chan_sip when automon in features.conf is enabled
ASTERISK-18806: Asterisk stops responding to SIP requests after DOS attack
ASTERISK-18807: [patch] pbx.c silently allows duplicate labels for the same extension, and shouldn't. Suggested [patch] included!
ASTERISK-18809: pbx_config.c assumes [macro-stdexten]
ASTERISK-18810: Function QUEUE_MEMBER broken after module reload
ASTERISK-18811: Dialplan is not processed after AGI script when dst channel hangup first
ASTERISK-18812: Wrong CallerID
ASTERISK-18813: Fix misleading gcc warning messages
ASTERISK-18823: TestSuite: Fix the Asterisk wrapper class to better manage an instance of Asterisk
ASTERISK-18824: CLONE - TestSuite: Fix the Asterisk wrapper class to better manage an instance of Asterisk
ASTERISK-18826: Blind Transfer failure
ASTERISK-18827: iax2 peer/trunk unreachable
ASTERISK-18828: CEL RADIUS garbage in attribute values
ASTERISK-18829: ConfBridge deadlocks waiting endlessly for a condition to be signalled inside bridge_channel_join_multithreaded
ASTERISK-18833: CLONE -Fix misleading gcc warning messages
ASTERISK-18835: res_monitor causing deadlock with no calls coming through
ASTERISK-18836: Crash caused by destruction of libc memory pool by destruction of internal asterisk structures: CDR variables.
ASTERISK-18837: empty Sender-Adress if use TCP-Protocoll for SIP-Messages
ASTERISK-18838: app_voicemail [general] variables zonetag and locale are not set on mailbox until after reload
ASTERISK-18839: [patch] missing unlock in sip_send_mwi_to_peer causes deadlock
ASTERISK-18840: double free or corruption crash with musiconhold
ASTERISK-18841: Call progress does not work with analog DAHDI cards
ASTERISK-18842: text config files are treated as unchanged when changing and reloading them quickly one after another
ASTERISK-18843: moh/loop mp3 files (mpg123) stop playing after core reload when using res_timing_dahdi
ASTERISK-18844: Extension state callback needs to happen when callback is removed.
ASTERISK-18845: chan_gtalk only receives calls from Gtalk Android but not viceversa
ASTERISK-18846: Asterisk crash on issuing Asterisk-CLI 'reload' command multiple times
ASTERISK-18847: Asterisk 10.0.0 Release Blockers
ASTERISK-18848: Moduleinfo section in app_macro.c is not terminated properly
ASTERISK-18849: Add support for video codec media attributes
ASTERISK-18852: mxml puts quotes inside multiline opaque data
ASTERISK-18857: misspelling in main/pbx.c
ASTERISK-18859: Unable to place calls from some SIP phones ("Multiple audio streams are not supported")
ASTERISK-18862: Change default for sip.conf 'nat' setting
ASTERISK-18863: Call hang-up when issuing "mixmonitor start" just after "bridge" through AMI
ASTERISK-18867: Incorrect Password in jabber.conf leads to memory leak
ASTERISK-18868: sig_pri.c references an incorrect variable inside restart event handler
ASTERISK-18871: Problem by enabling simultaneous two spans in MFC R2
ASTERISK-18873: MEETME_RECORDINGFILE only read when realtime meetme conference is first loaded from the database
ASTERISK-18875: CDR logs doesn't take care of CDR(dest) variable
ASTERISK-18876: When g729 codec is configured with packetization time larger than 120
ASTERISK-18879: Memory leak inside cel_pgsql
ASTERISK-18880: CLONE -Memory leak inside cel_pgsql
ASTERISK-18882: Asterisk lock during production
ASTERISK-18883: Asterisk TestSuite - test SIP/realtime_sipregs seg faults on exit
ASTERISK-18885: confbridge, video hangs, Exceptionally long queue length queuing to Bridge
ASTERISK-18886: database show fails for FOO/BAR/BAZ with sqlite where BDB succeeds
ASTERISK-18887: Asterisk doesn't respect the codec order - alaw always first in realtime.
ASTERISK-18889: SRTP packet corruption with SRTCP packet contents
ASTERISK-18892: CLONE -Asterisk doesn't respect the codec order - alaw always first in realtime.
ASTERISK-18895: ConfBridge application does not read sound_only_one config variable
ASTERISK-18897: [patch] SIP Notify Request without "Voice-Message" in the body is not accepted.
ASTERISK-18899: Erroneous ISDN 44 Rejection Hangup() bug
ASTERISK-18901: 1.4 app_read.c drops user out during read with no warning
ASTERISK-18903: Asterisk 10 RC2 Drops To Address from SIP Message
ASTERISK-18904: Parking timeout does not go to parkedcallstimeout
ASTERISK-18906: Add manager event on out of call message from SIP
ASTERISK-18907: AMI crashes on certain commands
ASTERISK-18908: Meetme - Use voicemail "greet" soundfile for user announce
ASTERISK-18909: Infinite loop in dialplan pattern parsing
ASTERISK-18911: When you turn on "sip set debug peer blah" it enables all sip debugging prompts for all SIP channels
ASTERISK-18912: Realtime MOH with caching plays a new song for every new hold within a call
ASTERISK-18913: segfault in cdr_adaptive_odbc.c when database connection is interrupted
ASTERISK-18914: NAT option doesn't show in sip peer's settings
ASTERISK-18915: Crash on duplicate free in chan_iax2 scheduler
ASTERISK-18916: Asterisk 10 RC2 Returns 484 Address Incomplete After latest SIP Message fix deployed
ASTERISK-18917: Asterisk 10 Incorrectly Formats From Header
ASTERISK-18918: Macro Exit via "Goto()" function, add support for text/named extensions
ASTERISK-18919: SIP MESSAGE body is not used verbatim
ASTERISK-18920: Silence after attended transfer on ring
ASTERISK-18921: sendfax_exec clears LOCALSTATIONID before sending fax
ASTERISK-18922: crash when you call 'core show channel <channel's name>'
ASTERISK-18923: res_fax_spandsp usage counter is wrong
ASTERISK-18924: Linksys devices SIP INFO messages - dtmf-relay signal value uses 0-9, #, *, a-d. Asterisk looking for 0-9, #, *, A-D
ASTERISK-18925: Asterisk sends "183 Ringing" in sipfrag bodies
ASTERISK-18926: 10.0.0-rc2 compiles, but chan_sip.so and chan_iax2.so can`t be loaded after compile
ASTERISK-18927: CLIP India
ASTERISK-18928: Implicit Assumption About Dynamic Features
ASTERISK-18929: main/asterisk.c compile error on OpenBSD
ASTERISK-18930: Asterisk stops responding to SIP devices if it loses Internet Access (DNS)
ASTERISK-18937: CLONE - Read factory 0xb6d0acb8 was pretty quick last time, waiting for them
ASTERISK-18938: Park() does auto-answer midway
ASTERISK-18939: CLONE - [regression] LIMIT_CONNECT_FILE does not get played to caller when using dial() app
ASTERISK-18940: Channel SIP do not get answer
ASTERISK-18941: Update Asterisk versions Wiki page with feature-freeze dates
ASTERISK-18943: Include iLBC source code for distribution with Asterisk
ASTERISK-18945: Review and refine the 'ignorebusy' option in app_queue
ASTERISK-18947: Document LINKKEDID_END event
ASTERISK-18948: core show channels randomly shows IP instead of IAX account
ASTERISK-18949: Segmentation fault in chan_sip.c (SIP/TLS SRTP configuration)
ASTERISK-18950: weak (linker) attribute handling for MAC/OS in optional_api.h breaks x86 executable (seg fault)
ASTERISK-18951: [regression] T.38 pass through produce 100% CPU usage spike
ASTERISK-18953: Sometimes bridge action fails
ASTERISK-18955: Voicemail message recording from IAX source sped up and jittery
ASTERISK-18956: autocreatepeer enhancement (add prefix option for safer peers)
ASTERISK-18957: asterisk-core-sounds-ru: LICENSE file missing
ASTERISK-18958: Asterisk Manager incorrectly sets a ChannelID to be global in highly repeatable circumstances
ASTERISK-18959: astdb2sqlite3 fails to run if it is missing from PATH
ASTERISK-18961: make fails on cross-compiling for ARM (armVFP)
ASTERISK-18962: The in CLI documentation for SayNumber is wrong
ASTERISK-18963: cel_sqlite3_custom creates table
ASTERISK-18964: Stuttering jittery audio after MOH
ASTERISK-18966: Consistent AMI error causing Channel variable setting to create global variables
ASTERISK-18967: [SIP] nonceCaching
ASTERISK-18968: CALLERID(num) do not work if the fromuser is set ?
ASTERISK-18969: Followme does not handle inital Connected Line updates.
ASTERISK-18970: Calls to undefined extensions are not logged into CEL
ASTERISK-18971: Inbound Gtalk calls fail randomly
ASTERISK-18973: Asterisk core dumps in video
ASTERISK-18974: trunk: unable to run unit tests
ASTERISK-18975: Manager Redirect action on bridged channel pair causes intermittent hangup on second channel
ASTERISK-18976: pbx_lua and confbridge menu dialplan_exec() do not work together
ASTERISK-18977: play announcement between music-on-hold files
ASTERISK-18978: Australia Accented audio files for the conference bridge rewrite.
ASTERISK-18979: Segmentation fault in scheduled event - send_provisional_keepalive_full
ASTERISK-18986: Segmentation fault when cdr_mysql.conf file contains errors
ASTERISK-18987: Alcatel workaround broke EARLY MEDIA for BeroFix
ASTERISK-18988: Confbridge ghost channels, segmentation fault under high load
ASTERISK-18989: Dropout in Moh. Seems to be a res_timing_timerfd issue
ASTERISK-18990: After upgrade from 1.6 to 1.8 one side audio in SPA941
ASTERISK-18991: CLONE - Channel SIP do not get answer
ASTERISK-18992: Asterisk From and To fields setup for SIP out of dialog MESSAGE method
ASTERISK-18993: CLONE - Flooding with chan_sip.c:6047 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x
ASTERISK-18994: playback of file format without seeking is broken
ASTERISK-18995: Support for OGG/Speex file format
ASTERISK-18996: Faulty SIP session timer handling
ASTERISK-18997: Segfault in res_config_odbc/res_odbc when using realtime peers (sipregs)
ASTERISK-18999: Connected Line updates need a disable option on a per SIP device / trunk basis.